[Libav-user] AudioToolbox AudioBufferList to/from AVFilter

[Ronak]

Hi Paul,

Thanks for that, that fixed the problem. Now, I'm getting -35 errors when I try to read the audio out of the buffer sink:

Why would that happen?

- (void)allocateRenderResourcesAudioFormat:(AVAudioFormat * _Nonnull)format capacity:(AVAudioFrameCount __unused)frameCapacity {

  NSString *bufferArgs = [[NSString alloc] initWithFormat:@"sample_rate=%f:channels=%d:sample_fmt=%s:channel_layout=%d", format.sampleRate, format.channelCount, av_get_sample_fmt_name(AV_SAMPLE_FMT_FLTP), AV_CH_LAYOUT_STEREO];

  avfilter_graph_create_filter(&_bufferContext, avfilter_get_by_name("abuffer"), "buffer_context", bufferArgs.UTF8String, NULL, self.filterGraph);
  avfilter_graph_create_filter(&_bufferSinkContext, avfilter_get_by_name("abuffersink"), "buffer_sink", NULL, NULL, self.filterGraph);

  avfilter_graph_create_filter(&_bassFilterContext, avfilter_get_by_name("bass"), "bass", "gain=0:frequency=100:width_type=o:width=1", NULL, self.filterGraph);
  avfilter_graph_create_filter(&_trebleFilterContext, avfilter_get_by_name("treble"), "treble", "gain=0:frequency=10000:width_type=o:width=1", NULL, self.filterGraph);
  avfilter_graph_create_filter(&_equalizerFilterContext, avfilter_get_by_name("equalizer"), "equalizer", "gain=0:frequency=250:width_type=o:width=1", NULL, self.filterGraph);

  avfilter_link(_bufferContext, 0, _bassFilterContext, 0);
  avfilter_link(_bassFilterContext, 0, _trebleFilterContext, 0);
  avfilter_link(_trebleFilterContext, 0, _equalizerFilterContext, 0);
  avfilter_link(_equalizerFilterContext, 0, _bufferSinkContext, 0);

  avfilter_graph_config(self.filterGraph, NULL);
}

- (void)processBuffer:(AudioBufferList * _Nonnull)buffer outputBuffer:(AudioBufferList * _Nonnull)outputBuffer {

  AVFrame *audioFrame = av_frame_alloc();
  audioFrame->channels = 2;
  audioFrame->channel_layout = AV_CH_LAYOUT_STEREO;
  audioFrame->sample_rate = 44100.000000;
  audioFrame->format = AV_SAMPLE_FMT_FLTP;
  audioFrame->nb_samples = buffer->mBuffers[0].mDataByteSize/ sizeof(Float32) * 44100;
  audioFrame->pts = audioFrame->nb_samples;

  audioFrame->extended_data[0] = buffer->mBuffers[0].mData;
  audioFrame->extended_data[1] = buffer->mBuffers[1].mData;
  audioFrame->linesize[0] = buffer->mBuffers[0].mDataByteSize;

  int result = av_buffersrc_write_frame(self.bufferContext, audioFrame);
  if (result > 0) {
    AVFrame *returnedFrame = av_frame_alloc();
    int result3 = av_buffersink_get_frame(self.bufferSinkContext, returnedFrame);

    NSString *string = [[NSString alloc] initWithCString:av_err2str(result3) encoding:NSUTF8StringEncoding]; <---- This shows a -35 error code
    NSLog(@"The string is %@", string);

    outputBuffer->mBuffers[0].mData = returnedFrame->extended_data[0];
    outputBuffer->mBuffers[1].mData = returnedFrame->extended_data[1];
  } else {
    NSString *string = [[NSString alloc] initWithCString:av_err2str(result) encoding:NSUTF8StringEncoding];
    NSLog(@"The string is %@", string);

    outputBuffer->mBuffers[0].mData = buffer->mBuffers[0].mData;
    outputBuffer->mBuffers[1].mData = buffer->mBuffers[1].mData;
  }
}

Is there something wrong with the frame I'm passing into the call to av_buffersink_get_frame?


This is just something simple I'm trying to get up and running, before I write production level code.

Thanks for the help!

Ronak

> On Nov 28, 2017, at 11:58 AM, Paul B Mahol <onemda at gmail.com> wrote:
> 
> On 11/28/17, Ronak <ronak2121-at-yahoo.com at ffmpeg.org> wrote:
>> I managed to trace this down to av_frame_get_buffer returning -22.
>> 
>> Here's the code that I tried:
>> 
>>  AVFrame *audioFrame = av_frame_alloc();
>>  audioFrame->channels = 2;
>>  audioFrame->channel_layout = av_get_default_channel_layout(2);
>>  audioFrame->sample_rate = 44100;
>>  audioFrame->nb_samples = buffer->mBuffers[0].mDataByteSize/
>> sizeof(Float32) * 44100;
>>  audioFrame->pts = audioFrame->nb_samples;
>>  av_frame_get_buffer(audioFrame, 0); <--- returns -22
> 
> You nowhere set sample format.
> 
>> 
>>  audioFrame->extended_data[0] = buffer->mBuffers[0].mData;
>>  audioFrame->extended_data[1] = buffer->mBuffers[1].mData;
>>  audioFrame->linesize[0] = buffer->mBuffers[0].mDataByteSize;
>> 
>>  AVFrame *otherFrame = av_frame_alloc();
>>  int result2 = av_frame_ref(otherFrame, audioFrame); <--- returns -22
>> 
>>  int result = av_buffersrc_write_frame(self.bufferContext, audioFrame);
>> 
>> Why would av_frame_get_buffer return -22? Am I not supposed to call it? What
>> about write frame?
>> 
>> 
>>> On Nov 27, 2017, at 7:19 PM, Ronak Patel
>>> <ronak2121-at-yahoo.com at ffmpeg.org> wrote:
>>> 
>>> Hi Paul,
>>> 
>>> Do you mind pointing me to the relevant documentation?
>>> 
>>> I tried setting up an AVFrame instance with the sample rate, channel
>>> layout and data but the calls to av_frame_ref are failing with -22 errors.
>>> I'm looking for any sample code that shows how to properly initialize an
>>> AVFrame from an AudioBufferList.
>>> 
>>> Thanks
>>> 
>>> Ronak
>>> 
>>> Sent from my iPhone
>>> 
>>>> On Nov 26, 2017, at 2:17 PM, Paul B Mahol <onemda at gmail.com> wrote:
>>>> 
>>>>> On 11/26/17, Ronak <ronak2121-at-yahoo.com at ffmpeg.org> wrote:
>>>>> Hi,
>>>>> 
>>>>> I'm trying to build a graphic equalizer using the ffmpeg library for
>>>>> iOS,
>>>>> wrapping the AVFilter library in an AUAudioUnit.
>>>>> 
>>>>> I'm having trouble figuring out how to convert an AudioBufferList's data
>>>>> to
>>>>> an AVFilter and back. The input buffers are in stereo, so I'm also
>>>>> unsure
>>>>> how to pass in both data arrays.
>>>>> 
>>>>> Does anyone know how to do this?
>>>> 
>>>> Have you read already available documentation?
>>>> 
>>>> AVFrame stores samples for packed format into AVFrame->data[0].
>>>> And planar format into AVFrame->extended_data[ X ], where X is channel
>>>> number.
>>>> _______________________________________________
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>>> 
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>> 
>> 
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